IP packet ready PBX expansion circuit for a conventional personal computer with expandable, distributed DSP architecture

ABSTRACT

A personal computer includes a PBX control program and one or more switch cards located in expansion board slots. The one or more switch cards are coupled to one or more port expansion units (PEU), which are coupled to telecommunication lines out to the various extension phones at the customer premises. Each PEU contains its own digital signal processor (DSP) so that distributed digital signal processing may be implemented to avoid any bottlenecks. One master PEU is coupled to a switch card by a time division multiplexed (TDMA) bus as well as a packet switched control bus, and all the other PEUs, if any, are coupled to the master PEU by extensions of the TDMA and packet switched buses. The TDMA bus carries PBX real time conversations while the packet switched bus carries control information, voicemail outbound message data packets, and inbound voicemail data packets.

FIELD OF USE

The invention pertains to the field of business telephone systems with PBX capability, and, more particularly to systems that can convert conventional personal computers into machines with PBX capability.

One of the problems with PBX on an expansion card technology for personal computers is lack of expandability to add new ports above and beyond the ports provided by the expansion card. Other problems with prior systems such as the Netphone and Altigen systems (Netphone's system is described in U.S. Pat. No. 5,659,005) are bottlenecks and latency caused by the use of a single centralized microprocessor to perform voice mail processing, outgoing message generation and other duties for all ports coupled to the card. The Netphone patent does teach time division multiplexed expansion interface 70 which can be used to couple multiple PBX cards like that shown in FIG. 2 together by a TDMA PCM highway. This expansion capability allows multiple PBX cards to be coupled together by a TDMA bus. The main PBX card is coupled to the server by an ISA bus interface 60. An ISA bus is a slow 16-bit, master-slave type system bus used on older personal computers such as the IBM AT. An ISA bus can only transfer one word at a time between the PBX card and the host server. It is not particularly well adapted for bursting large amounts of data over a short time such as is required to playback or record voicemail recordings.

Also, any host bus requires a bus master to drive the command and address lines of the bus to control which slave device is allowed to use the bus to communicate with the master and whether the communication will be a read or write transaction. A bus slave such as the PBX card cannot control the bus and must request control of the bus such as by asserting an interrupt request line on the system bus. An interrupt request requires a context switch by the host server wherein it stores the state of all of its internal registers and program counter and vectors to an interrupt service routine which takes control of the bus and allows the PBX card to send data to the host or receive data from the host. Each such context switch causes delay.

There is no memory on the Netphone PBX card which is large enough to be capable of recording digitized voice of voicemail or IP telephony packets or other multimedia streams data such as MP3 recordings coming in from the internet via a T1 line connection. Thus, all this data that needs to be recorded must be sent to the server for recording on the host hard disk. The ISA bus will be a bottleneck in this situation and could lead to loss of data if the RAM 64 overflows.

Further, even if multiple PBX cards are coupled together in the Netphone system, there is only one time division multiplexed bus that couples the cards together. This single TDM bus must be shared between carrying digitized voice of PBX connections between extension phones and one or more CO trunk lines or between extensions as well as outgoing or incoming voicemail data or IP telephony or multimedia data from the internet. The bandwidth requirements can exceed the available bandwidth of the bus and cause latency or lost data. Also, since voicemail recording or playback data, T1 type or IP telephony data and multimedia data is bursty traffic whereas TDMA buses are isochronous, there is a mismatch in the nature of the traffic versus the nature of the bus available to carry it. This can cause the need for additional buffer to hold the overflow thereby adding cost and delay to the process of transferring this bursty traffic from one port to another or from a port to the host server.

Another problem with current PBX systems that are built around personal computers is that in order to provide PBX connections to a T1, DSL or ISDN digital service, an expansion card which uses up one of the precious expansion slots on the host motherboard must be used. It is disadvantageous for any PBX system to make connections to CO POTS lines or digital lines from the CO or extension lines out to extension phones (hereafter “ports”) through expansion cards that couple to expansion slots in the motherboard. This is because there are only a limited number of expansion slots on the host motherboard, and once they are used up, the PBX system has gotten as large as it can get and no more ports can be added without modifications to the host motherboard to add more expansion slots which may render it inoperative or slow down the system bus by loading it with parasitic capacitive and inductive loads higher than it was designed to handle for a given bit rate and clock speed.

Thus, a need has arisen for PBX system wherein a large, almost unlimited number of ports may be added in small slices by the addition of individual port expansion units to a standard, unmodified inexpensive personal computer where the port expansion units do not need to be coupled to expansion slots on the motherboard. A need has further arisen for a PBX system built around a conventional, inexpensive personal computer where the PEUs can interface either to POTS CO lines, any type of digital service to the CO or to extension lines coupled to extension phones. A need has further arisen for a PBX system built around a conventional, inexpensive personal computer where the PEUs interface to the host server through both a TDMA bus for PBX connection traffic of real time conversations and a packet switched bus carrying control and voicemail data which can easily handle the bursty and packetized nature of voicemail, ISDN, DSL and IP telephony packet-based traffic.

SUMMARY OF THE INVENTION

The genus of the invention is a class of PBX machines built around a standard, unmodified personal computer programmed with a PBX control program and to which is added one or more switch cards in expansion bus slots coupled to one or more port expansion units (PEU) which are not coupled to bus expansion slots and which couple the PBX to POTS or DSL or ISDN trunk lines or extension lines out to the various extension phones at the customer premises. Each PEU contains its own digital signal processor (DSP) so that distributed digital signal processing may be implemented to avoid any bottlenecks. One master PEU is coupled to a switch card by a time division multiplexed (TDMA) bus as well as a packet switched control bus, and all the other PEUs, if any, are coupled to the master PEU by extensions of the TDMA and packet switched buses. The TDMA bus is used to carry port-to-port PBX real time conversations while the packet switched bus is used to carry control information, any type of packetized digital data, voicemail outbound message data packets, and inbound voicemail data packets to be recorded on the hard disk of the server. Each port expansion unit has multiple ports which can be coupled to one or more central office trunk lines or one or more lines to extension telephones. Some port expansion units (PEU) are designed to couple to T1 lines, ISDN lines or DSL lines.

The distributed DSP architecture allows a very large number of ports to be serviced and a large number of simultaneous voicemail or IP or other digital packet switched channels to be serviced without overwhelming the digital signal processing capability of the system. This is because there is a DSP in each PEU which handles processing necessary for voicemail playback and recording and any other digital signals processing necessary to support the analog or digital traffic entering and leaving its ports. Each switch card coupled to the host server motherboard also has its own DSP to handle conference calling and, in some embodiments, other processing in support of the centralized switching function. The switch cards are coupled to each other by a high capacity TDMA bus so that PBX real time conversations can be routed from one switch card to another over the TDMA bus and do not have to load down the host system bus, which is typically a PCI bus in the current generation of Pentium/Celeron class host personal computers.

The use of separate TDMA and packet switched buses to couple each PEU to the switch cards in the server allows a large number of voice mail channels to be simultaneously serviced via the packet switched bus with low or no latency without any impact on the capacity to carry PBX-connected, real time conversations on the TDMA bus coupling the PEUs to the switch cards. Further, the packet switched bus is better suited than a host bus or a TDMA bus to handle packet based bursty traffic such as is found in T1 packet based TDMA traffic, or ISDN or DSL or IP telephony packet based traffic as multimedia stream traffic.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is an overall block diagram of a PBX system according to the teachings of the invention;

FIG. 2 illustrates a time division multiplexed expansion interface to couple multiple PBX cards together according to an embodiment of the invention;

FIG. 3 illustrates a format of packets transmitted back and forth on a packet bus according to an embodiment of the present invention;

FIG. 4 is a diagram of a polling token according to an embodiment of the invention;

FIG. 5 illustrates a block diagram of PEU interface logic according to an embodiment of the invention;

FIG. 6 illustrates a block diagram of a PEU's telephone interface circuit according to an embodiment of the invention;

FIGS. 7A-7D illustrate a flowchart for processing that occurs in a PEU by DSPs and a microcontroller to carry out the commands of PBX call control and driver processes running on a host according to an embodiment of the present invention;

FIGS. 8A and 8B illustrate a flowchart of a process an MC carries out to interface with telephone line interface circuits according to an embodiment of the invention;

FIGS. 9A AND 9B illustrate a flowchart of PEU processing for an outgoing phone call placed from an extension coupled to a PBX system according to an embodiment of the invention;

FIGS. 10A and 10B illustrate a flowchart of processing of a TEU for incoming T1 calls according to an embodiment of the invention;

FIGS. 11A AND 11B illustrate a flowchart of processing for placement of an outbound call on a T1 port according to an embodiment of the invention; and

FIG. 12 is a flowchart of processing of DSPs on switch cards according to an embodiment of the invention.

DETAILED DESCRIPTION OF INVENTION

Distributed Multicard DSP Architecture With TDMA and Packet Switched Buses Coupling PEU Cards to Switch Cards

Referring to FIG. 1, there is shown a block diagram of an easily expandable, highly capable PBX system built around a conventional, unmodified personal computer 10 which can interface not only to analog POTS lines but to digital services such as T1, ISDN or DSL as well. The personal computer 10 can be any type personal computer such as a Windows based Pentium/Celeron class machine or a Power Macintosh class machine. Slower machines with less capable operating systems and less spacious hard drives could also be used with a resulting loss in system speed. In the preferred embodiment, the conventional personal computer is a Pentium/Celeron class microprocessor 14 running the Windows NT operating system and communicating with a PCI system bus 12. Not all the components of the personal computer are shown, but some of the more important components are shown.

A PBX control program is stored on hard disk 16 and controls the CPU 14 to control the various other components of the system to implement at least PBX functionality. Any PBX control program which can control the switch card to do the proper switching and which uses the TDMA bus to the PEU cards for real time PBX voice connections and uses the packet switched bus to send and receive control packets will suffice to practice the invention. Optionally, the control program also uses the packet switched bus to send and receive packets containing prerecorded voice messages, voicemail, IP telephony, and other digital data traffic from ISDN, T1 or DSL lines to and from the PEU cards. These other digital services implement such things as wide-band internet access, NetMeeting traffic, multimedia stream data etc. Optionally, the control program distributes those packets on a local area network 18 via a network interface card 20.

In the preferred embodiment, server 10 is coupled via any type local area network 18 to one or more client computers, of which computers 22, 24 and 26 are typical. The PBX software on host 14 communicates with these telephony enabled client applications via the TAPI interface. In alternative embodiments, the server 10 is not coupled to any client computers by a local area network and is only coupled to central office POTS or T1 or DSL or ISDN or other digital service lines and to lines out to one or more extension phones. In the preferred embodiment, the client computers are programmed with telephony enabled applications which can be used to send commands over the LAN to control switching, dialing, or other PBX functionality provided by the server such as sharing connections to a T1 line or sharing an ISDN or DSL connection or sharing a CO POTS trunk line between data/FAX modems in each client computer.

The computer 10 is modified to provide basic PBX switching capability by the addition of one or more switch cards 28, 30 and 32. Each switch card can drive two chains of PEU cards to which it is coupled. Each switch card has two connectors for each chain. One connector is for a TDMA bus like bus 50 which has 128 channels or timeslots per frame with an 8 KHz frame rate. The other connector is for a packet switched bus like bus 54. Each switch card is coupled to the other switch cards by a time division multiplexed bus 34.

In the preferred embodiment, this bus 34 follows the ICTF H.100 standard, which is hereby incorporated by reference, but any TDMA bus with sufficient bandwidth for the possible maximum traffic volume will suffice. The H.100 bus has 32 lines each of which has a bit clock rate of 8.192 MHz with 128 timeslots per line per frame and 8 bits per timeslot and a frame rate of 8000 frames per second (matching the T1 frame rate and the frame rate on buses 50 and 126 and matching the timeslot size on the T1 lines and buses 50 and 126) for a data rate on each line of 64 KBPS which is the necessary data rate for digitized voice. Over 32 lines, this provides 4096 available channels per frame that can carry data from one port to another. Hereafter, because there are several TDM buses in the system, bus 34 may be referred to as the H.100 bus.

In the preferred embodiment, each timeslot is one channel having 64 KBPS bandwidth. This is not enough for compressed video which typically requires at least 2 MBPS. However, in alternative embodiments, bigger timeslots could be used to obtain the needed bandwidth to carry compressed video data such as video conferencing data or video on demand data. In other alternative embodiments, groups of smaller timeslots all carrying compressed video data from a single program can be switched together to provide the necessary bandwidth. Typically, the ports 58, 60 etc. are mapped sequentially at startup time to corresponding single channels of the 256 channels through the TDM switch circuit 36 (the 256 channels through the switch card are split between two 128 timeslot/frame TDMA buses like bus 50). Any T1 port is mapped to 24 consecutive timeslots to match the 24 timeslots of the T1 line. The data rate of each T1 timeslot is 64 KBPS so the data from each T1 timeslot can be carried by a single channel on the TDMA bus 50.

Obviously, in a small system with only 10-15 ports, most of the bandwidth on the TDMA bus 50 is wasted. Thus, in small systems, there is bandwidth to spare on the TDMA bus, and an alternative embodiment is to use the unused channels in blocks which are large enough to support compressed video and switch all the blocks of channels together. This requires a modification of the PBX control software controlling host 14 to write control bits into the control registers of the TDM switch 36 appropriately such that channels that are all carrying parts of the compressed video of a single program are switched together. Likewise for the T1/ISDN interface 98 and DSP 100 of the PEU card 70. T1 lines carry 24 timeslots in each direction each of which carry 8 bits. If, for example, six timeslots are blocked together as a superchannel to carry incoming compressed video of a single video conference, all six timeslots will be switched as a block and mapped onto six corresponding timeslots of TDMA bus 50 by DSP 100.

In the preferred embodiment, each switch card is also coupled to the host microprocessor 14 by the system bus 12 and an expansion slot through a conventional PCI bus interface chipset 38. The host can be any personal computer with sufficient disk space and clock speed to handle the expected volume of traffic, and this depends upon the number of ports and the expected volume of voicemail traffic if voicemail is implemented. A PCI bus based computer is preferred, and there are some Power Macs that have PCI buses, but in alternative embodiments, other buses, host computers and operating systems may also be used so long as the PBX and any other driver software for the specific switch card and PEU designs in use are ported to the host's operating system.

Each switch card includes a TDM switching circuit 36 such as an OKI Semiconductor 53812-Model CTB12 chipset which drives the H.100 bus. This switching circuit is controlled by the PBX application software through messages passed to it through the operating system and the system bus 12 or by memory mapping of I/O register bits on the switch cards I/O registers into the I/O memory space of the host 14 with each bit having predefined functions for its on and off state, or by any other known mechanism for transferring control information to hardware and/or software processes running on the switch card. In the preferred embodiment, control bits in registers on each switch card are read by the host to determine status of the switch cards. Likewise, the host controls switching and other functions of the switch card by writing these bits.

The TDM switching circuit 36 on each switch card does switching of data between timeslots on the H.100 bus 34 in the following way. Each switching circuit 36 has two sets of 256 registers therein. If the timeslots on the H.100 bus are numbered 0 through 4095, the host 14 will write a number from 0 to 4095 into a control register on the switch card and that is how the host controls the PBX connections that are made through specific timeslots. The switching circuit logically functions as if there were 256 multiplexers for each direction of data travel. Each of these 256 multiplexers represents one channel. Each of these 256 logical multiplexers has 4096 inputs each receiving the data from one of the 4096 timeslots on the H.100 bus. Each multiplexer has one output line at which the data from a selected one of the 4096 input timeslots appears, and that data will be the data for that channel for that direction of travel. Each of the 256 multiplexers has its switching control input coupled to one of the 256 registers. The number written into the register by the host controls which timeslot's data is coupled to the output.

TDMA data multiplexing requires synchronization, i.e., frame alignment by the transmitter and receiver so each is synchronized to put data into and take data out of the correct timeslots. The H.100 bus includes a control line which carries a frame sync signal which indicates the occurrence of the beginning of timeslot zero so all circuits coupled to the H.100 bus can stay in synchronization.

The selected 256 timeslots of data traveling in the direction away from the switch 36 are output on a TDMA bus 40 for input to a digital signal processor (DSP) 42. The 256 timeslots of data traveling from the ports and the DSP 42 toward the switch are output by the DSP on “local” TDMA bus 40. The term ports will be used herein to refer to the external ports coupled to central office POTS lines, extension phones, T1, ISDN or DSL lines. Ports such as 58 and 60 and 62 are typical. The 256 timeslots traveling in each direction on local TDMA bus 40 will be referred to as the local timeslots.

The overall function of the switching circuit is to provide switchable data paths at least between ports to implement at least PBX connectivity under the control of the PBX application software.

Virtual Conference Rooms

The DSP 42 is programmed to implement conference calling so that “virtual conference rooms” can be set up in the phone system. This is done by routing the data from all the timeslots of each participant in the conference to each of the ports to which a participant is coupled.

The DSP is coupled by another local TDMA bus 44 to an external interface FPGA 46. The external interface 46 is shown in block diagram form as FIG. 3. The external interface has buffers 64 to buffer the timeslot data traveling in both directions to and from the PEU card 52 on TDMA extension buses 40, 44, 48 and 50. There is an FPGA interface circuit 46 on each switch card and an FPGA interface circuit 104 on each PEU. The functionality of each of these FPGA types is described below in VHDL language in Appendices A and B.

The PEU 66 is comprised of one or more port expansion unit circuit boards (hereafter referred to as a card) of which cards 66, 70, 72 and 74 are typical. Each PEU card has one or more ports for coupling to CO POTS trunk lines, T1, ISDN or DSL lines or lines to extension phones. The PEU cards do not use up expansion slots in the host computer 68. Each PEU card is coupled to its neighboring PEU cards by a TDMA bus 50 and a packet switched bus 54 to form a daisy chain. The first PEU in the chain is coupled to the server 68 by a TDMA bus 50 and a packet switched bus 54. The significance of this architecture is that the system can be easily expanded to grow to as many extension phones and CO POTS trunk lines and T1 or DSL lines as needed to meet the needs of the customer while using a conventional personal computer with only a limited number of expansion slots as the server. In addition, it is new to have a packet switched bus interface on each PEU and it is also new to use the packet switched bus for voice mail traffic. Prior art PBX on a card type circuitry uses the TDMA path and the switching resource to couple the voice resources such as voice mail files stored on the hard disk to the codecs at the ports. Typically, only a fixed number of TDMA channels are devoted to voicemail traffic, so when those channels are all busy, the system has a bottleneck and other ports cannot play or record voicemail. This centralized limitation of voicemail channels to some fixed number severely limits the expandability of the system in terms of the number of extension telephones, T1, ISDN or DSL lines or CO POTS lines which can be coupled to the system.

The external interface 46 is preferably a field programmable gate array. It is coupled to the first PEU card in the chain via the TDM signal buffers 64 and the TDMA bus 50. Clock generation circuit and frame sync signal generator 108 generate the remaining two signals on the TDMA bus 50. The external interface 46 also includes an address generation circuit 71, an SRAM FIFO 74 and conventional transmitter and receiver circuits 76 and 78 the outputs of which form the packet switched bus 54.

The format of the packets transmitted back and forth on the packet bus 54 is shown in FIG. 3. Each packet includes a size field 80, a destination address 82, a source address 84, a type field 86 that contains data indicating whether it is voicemail data, a command or some other type of data such as an IP packet etc., a payload section 88 and CRC bits 90 for error detection and correction.

The host CPU 14 is coupled to the packet switched bus for sending and receiving messages via the host system bus 12, the bus interface circuit 38, the SRAM FIFO 74 and the transmitter 76 and receiver 78 circuits. The transmitter and receiver circuits function to receive control message packets, voicemail packets, multimedia stream packets, IP packets and IP telephony packets traveling in both directions between the PEU circuits, the host 14 and drive them onto or receive them from the packet switched bus 54.

The packet data interface between each PEU and the server 68 via the packet switched bus is implemented by a microcontroller on each PEU card, of which microcontrollers 92 and 94 are typical. The microcontroller on each PEU card also functions to control signaling on the telephony interface regardless of whether it is POTS or an extension line or a T1 or other digital port. The microcontroller also transfers packetized voice data back and forth in packet format over the packet switched bus 54 to play and record voice messages. The microcontroller on each PEU card also coordinates with the digital signal processor (hereafter DSP) on the PEU card to implement DTMF tone recognition and generation.

The function of the DSP on each PEU card is to convert packetized voice data received from the microcontroller to the TDMA format of the telephony interface used on a TDMA PEU bus in each PEU. The DSP on each PEU card also does tone recognition and generation, and also functions to implement a gain control. Because both the DSP and microcontroller on each PEU are software controlled, their roles and functionality are flexible and can be changed or expanded in the future. The clock speed and bandwidth of the local TDMA bus 96 on each PEU card is different and slower than the clock speed and bandwidth on the TDMA bus 50. The DSP 102 on each PEU card resynchronizes the data transfer between the buses using synchronization signals from the interface logic 104. Each TDMA bus in the system is comprised of four wires: a clock, a synchronization signal, an input data, and an output data wire. The clock speed on the local bus 96 is 2.048 MHz and there are 32 timeslots per frame each carrying 8 bits at a frame rate of 8000 frames per second. There are up to 16 ports per PEU card and up to 14 PEUs in the chain coupled to each switch card. The TDMA bus 50 between a switch card and the first PEU in its chain has 256 timeslots per frame and a clock speed of 8.192 MHz with 8 bits per timeslot and a frame rate of 8000 frames per second. The local bus 96 runs as a lower clock speed because the particular CODEC in the telephone interface 122 that digitizes the incoming audio signals from the port and which receives digital data from the timeslots on bus 96 and converts it to audio signals cannot run at a higher clock speed. However, in alternative embodiments where a faster CODEC is used, the local bus 96 can run at the same clock rate as the TDMA bus 50.

The DSP102 on each PEU resynchronizes the data coming in from the timeslots on TDMA bus 50 to the timing of the corresponding timeslots on the local TDMA bus 96 and vice versa. There are actually two DSPs per PEU card if the PEU has sixteen ports. Each DSP handles eight of the ports. The DSPs are synchronized to the timeslots on the local TDMA bus 96 by a synchronization signal received from the interface logic FPGA 104. The interface logic FPGA 104 synchronizes each DSPs with a sync pulse.

Each port is mapped at startup time to one timeslot on the local TDMA bus 96 and a corresponding timeslot on the TDMA bus 50. If a port is coupled to a T1 line, then 24 timeslots on the TDMA bus 50 are assigned to it. When the sync pulse arrives from the interface logic 104, the DSP knows that the next eight timeslots on the TDMA bus 96 belong to it. The data in and data out wires of TDMA buses 96 and 50 are frame synchronized, so when the DSP receives its sync pulse, it reads the data coming in from the eight ports assigned to these eight timeslots and writes data going out to these same eight ports into the timeslots assigned to those ports.

There are no collisions on the TDMA bus 50 because each of the ports on all the PEUs is assigned to one unique timeslot. A PEU coupled to a T1 line maps the 24 timeslots of the T1 data input line to 24 timeslots on the TDMA bus 50 and maps the 24 data output timeslots to 24 different timeslots on the TDMA bus 50.

However, some mechanism must be employed to prevent collisions on the packet switched bus 54 between transmissions from various PEU cards in the chain and the server. Any known mechanism to do this will suffice to practice the invention. In the preferred embodiment, a token based polling scheme is used. FIG. 4 is a diagram of the polling token. The token is comprised of a start bit, a 4-bit address, a match bit and a stop bit. There are 16 possible addresses of which 0 is illegal and 15 is broadcast. The address generator generates tokens with addresses 1-14 sequentially without any knowledge of how many PEU cards are actually connected in the preferred embodiment. In alternative embodiments, the address generation circuitry could periodically send out discovery tokens requiring a reply with the address of the PEU that replies to determine how many PEUs are on the bus and what their addresses are and then generate tokens sequentially in a round robin fashion directed to only the PEUs that are actually connected to the bus 54.

Each PEU watches the address line for the token, and when a token arrives, it compares the address in the token to its own address. If the address does not match, the token is transmitted to the next PEU in the chain on the address line of the packet bus 54. If the address does match, the PEU changes the match bit from 0 to 1 and retransmits the token. The next PEU, if it already has an address, will change the match bit back to 0 and retransmit the token. The PEU with the matching address then has full control of the packet bus 54 until a token arrives that has an address that does not match. Thus, each PEU gets control of the packet bus for the interval during which the token address matches its own address. During that interval, packets of data stored in a transmit buffer of the interface logic 104 are transmitted to the server using packets of the form shown in FIG. 3, and packets addressed to that PEU are transmitted from the server to that PEU.

The match bit is used by new PEUs that have just been plugged into the bus to learn and acquire the next sequential address on the bus. If for example, two PEUs are already plugged into the bus, they will have already acquired addresses 1 and 2. Suppose now a third PEU is plugged into the bus. The third PEU learns its address by watching the match bit of tokens it receives. When PEU receives a token with address 1, it changes the match bit from 0 to 1 and sends the token to PEU 2. PEU 2 changes the match bit back from 1 to 0 and retransmits the token to the new PEU. When the new PEU powers up, it defaults to invalid address 0 and never finds a token addressed to it. However, when a token with address 2 arrives at PEU 2, it changes the match bit from 0 to 1 and sends it along to PEU 3. When PEU 3 sees the match bit of 1 and the address of 2, it automatically determines that its address will be 3 and stores that in its memory for future comparisons to token address fields. Other schemes could also be used. For example, a switch on each PEU could be set with that PEU address as is done on SCSI buses so that each PEU has a different address. In another alternative embodiment, a discovery process by the server could be periodically carried out and the server could assign an address each time a new client PEU is found.

A block diagram of the PEU interface logic FPGA is shown in FIG. 5. The four lines of the TDM bus 50 are coupled to a TDM channel synchronization and selection circuit 124. Those four lines are a clock CLK, a frame synchronization signal line, FS, a transmit data line TD and a receive data line RD. The purpose of the sync and selection circuit 124 is to retransmit the incoming data from bus 50 to the DSP on another TDMA bus 126 but to generate a second frame sync signal FS#2 on line 128. Bus 126 is coupled to each of the DSPs on the PEU (if there are more than one) and has the same clock rate of 8.192 MHz as bus 50 and has a frame sync signal FS on line 130 that keeps the DSP 102 synchronized to the timeslots on the transmit data line 132 and the receive data line 134. The function of each DSP is to put the data from timeslots on bus 126 directed to particular ones of the ports the DSP is controlling onto the proper timeslot of the local bus 96. This will require some re-timing and synchronization from the synchronization and selection circuit 124. This synchronization function is performed by the frame sync signals FS#2 and FS#3 on lines 128 and 132. These frame sync signals are at the same 8 KHz frame rate as on the faster TDM buses 50 and 126 because the total duration of a 32 timeslot frame on local bus 96 is the same as the total frame duration of the 128 timeslots on TDM buses 50 and 126. However, the frame sync signals on lines 128 and 132 are offset in time from each other by ¼ of the total frame time on local bus 96. Line 128 is coupled to one of the DSPs coupled to the local bus 96 and line 132 is coupled to the other DSP coupled to the local bus 96. Each frame sync signal on lines 128 and 132 when it occurs tell the DSP to put received data onto the next 8 timeslots on the local bus 96 and to take data received during the next 8 timeslots on bus 96 and put it on timeslots of the receive data line 134 of bus 126.

Each DSP receives incoming data from the switch card on line 132 and stores it into its receive buffer and then processes it for volume control and writes it to its transmit buffer. The DSP also may write data for generated DTMF tones into the transmit buffer for the appropriate timeslot and port that needs these tones (such as for dialing numbers). The DSP also receives data from the ports and stores it in a receive buffer coupled to the local bus 96. That data is processed for volume control, DTMF tone recognition etc. and written into the appropriate portion of a transmit buffer for the bus 126.

In the case of a PEU which is a T1 interface, such as PEU 70, there are three DSPs per T1 port 62 and there are two T1 ports per PEU. The three DSPs each process the data from 8 of the 24 T1 timeslots to do volume control, DTMF tone generation and recognition. These three DSPs are daisy chained and only the first one is coupled to bus 126 of the synchronization and selection circuit 124. Only the last one interfaces via an 8.192 MHz TDMA bus 134 to the T1/ISDN interface chipset 98. In the preferred embodiment, the T1/ISDN interface (ISDN lines are also 24 timeslots per frame) is the DS2152 chipset commercially available from Dallas Semiconductor. Each of the 24 timeslots of data on TDM bus 50 mapped to the 24 T1 timeslots is output by PEU interface FPGA 138 in FIG. 1 on a 8.192 MHz TDM bus 140 similar to TDM bus 126 in FIG. 5. The 24 timeslots of data to be output to the T1 slots is written into the receive buffer of the first DSP 144 (only one DSP is shown in FIG. 1) and is then copied to its transmit buffer. The first DSP processes its 8 timeslots and then outputs this data to the receive buffer of the second DSP in the chain. The second DSP processes its 8 timeslots, respectively, and then copies the data of all 24 timeslots to its transmit buffer. The second DSP then outputs all 24 timeslots to the receive buffer of the third DSP in the chain. The third DSP then processes the data of its 8 timeslots and outputs the processes data to its transmit buffer. The third DSP then outputs all 24 timeslots of processed data to the T1/ISDN interface 98 via TDM bus 134.

Another function of the TDM channel synchronization and selection circuit 124 besides copying data from bus 50 and the timeslots assigned to the local ports onto bus 126 (or 140 for a T1 PEU), is to implement the daisy chain to the other PEUs. Each PEU is coupled to its neighboring PEUs in the chain by an extension of both the TDMA bus 50 and the packet switched bus 54. These extensions are shown as 50A, 50B, 50C. The extensions of the packet switched bus 54 are shown as 54A, 54B and 54C.

The circuitry of the PEU interface logic that controls the packet switched bus 54 is also shown in FIG. 5. An address decoder compares the tokens transmitted by the address generator 71 in FIG. 2 to the address of this particular PEU. If there is a match, the decoder activates the enable lines 150 and 152 thereby enabling the receiver 154 and transmitter 156, respectively. The transmitter 156 then transmits any packets stored in SRAM FIFO 158 by outputting them in serial format on line 162 at the bit rate established by packet bus clock CLK on line 160. A multiplexer 164 is enabled by a switch control command on line 166 to couple line 162 to line 168 anytime transmitter 156 is transmitting but to couple the receive data line 170 of packet bus extension 54A to line 168 at all other times. In alternative embodiments, the multiplexer 164 will select line 162 for coupling to line 168 at all times when the token is in possession of this PEU.

Simultaneously, incoming packets from the server on transmit data line 172 of the packet data bus are received and written into the receive data FIFO 158 via bus 178. FIFO 158 is actually two FIFO, one for packets traveling in each direction. The microcontroller 92 constantly polls the FIFO 158 for new received packets, but in alternative embodiments, the receiver can generate an interrupt to the microcontroller each time it writes a packet into the FIFO. When there is a packet in the FIFO for the microcontroller, it is read via the interface represented by line.

The address decoder 148 also re-transmits any received tokens out to the other PEUs on the chain via control line 176 of packet bus extension 54A.

Referring to FIG. 6, there is shown a block diagram of the PEU's telephone interface circuit 122 in FIG. 1. This interface will work for connection either to a trunk line, a conventional telephone or one of the proprietary phones sold by the assignee Picazo Communications, Inc. The trunk line etc is represented by line 180 and couples to an RJ11 connector 182. A conventional ring detection circuit detects the ring voltage on the line and sets a bit which can be read by the microcontroller via line 184 via a polling cycle. In alternative embodiments, the ring detection circuit can cause an interrupt to the microcontroller, which then executes an interrupt service routine to read the status of the ring detector.

A conventional tip and ring reversal circuit 186 functions to reverse the tip and ring lines when necessary to support an incoming direct inward dialing (DID) call. In DID calls, the caller dials a number that is mapped to a particular extension on the PBX system. The CO signals the PBX when it wishes to direct a DID call to the PBX by taking the CO line offhook (closing the tip and ring loop). The offhook condition is detected by a conventional loop current detection circuit 196. Ring signals are detected by a conventional ring detection circuit 214. The microcontroller polls (or retrieves in response to an interrupt service routine) the off hook status and sees, by the absence of ring status on line 184, that a DID call is incoming. This fact is sent to the host 14 by the microcontroller as a control packet. The host then sends back a control packet ordering the microcontroller to do a tip and ring reversal to signal the CO that the PBX is ready to receive the call. The PEU microcontroller then sends a REVERSE command via line 198 to the tip and ring reversal circuit 186 causing it to reverse the tip and ring voltage condition to +48 volts on the tip line relative to the ground reference on the ring line (as opposed to the normal −48 volts on tip relative to ring). In a DID call, the battery voltage or loop current feed is supplied by the PBX to the CO to signal the CO that the PBX is ready to receive the DID call. Each port is configured at the system initialization as to whether it will be coupled to a CO line or an extension line, and, if coupled to a CO line, whether the CO line supports DID. To support DID or when the port is coupled to an extension phone, it is necessary to supply the loop current (−48 volts on tip relative to ring) from the PBX, and that is the function of current feed circuit 200 and switch 202. Switch 202 is controlled by the microcontroller 92 on the basis of configuration data defining whether the port is coupled to an extension phone or supports either type of DID that requires the PBX to send loop current to the CO for signaling. The switch 202 is closed to connect the current feed circuit 200 to the 2-wire to 4-wire transformer when the port is coupled to an extension or supports DID. This causes the normal loop current to be supplied to line 180. For DID signaling, the tip and ring reversal circuit 186 can reverse the polarity of the voltage on the tip line relative to the ring line for DID signaling that the PBX is ready to receive the call.

A ground start signaling circuit 188 functions to implement the ground start signaling protocol and is switched into the circuit by switch 190. Switch 190 is set at initialization time by the microcontroller. The ground start signaling circuit detects ground start signals if switch 190 is set to couple it to pair 210. The status of signaling voltages on pair 210 can then be read by the ground start signaling circuit and the microcontroller can read this status by reading a status register therein via line 212 as part of a polling process or in response to an interrupt.

A conventional hook switch or ring signal circuit 192 is coupled to the microcontroller 92 via control line 194. Switch 206 is manually operated and exists for every port. It is set by the installer when the port is coupled to a CO line or an extension phone. One position of the switch indicates the port is coupled to a CO line. The other position indicates a connection to an extension phone. The microcontroller reads switch 206 for each port to determine whether the port is coupled to the CO or an extension phone. Circuit 192 performs an offhook function when line 180 is coupled to the CO When the microcontroller has determined line 180 is coupled to the CO and the microcontroller receives a control packet from the host 14 indicating the line 180 is to be taken offhook, a control signal is sent to the circuit 192 on line 194 causing it to force line 180 offhook. If pair 180 is coupled to an extension phone and microcontroller 92 receives a control packet containing a command to ring whatever phone is coupled to pair 180, the control signal on line 194 is activated and that causes circuit 192 to drive ring signal voltages onto pair 180.

When the phone coupled to line 180 goes offhook, this condition is detected by circuit 192 and can be read by the microcontroller via a polled read of a status register in loop current detection circuit 196 via line 208. The control lines and status lines shown in FIG. 6 may, in some embodiments, actually be the data bus of the microcontroller in some embodiments with the microcontroller address bus being used to individually address each circuit the status of which is to be read or to which a command is to be sent.

Analog signals from the CO or from an extension phone via pair 180 are digitized in CODEC 216 and put on local TDM bus 96 in the appropriate timeslots. A shift register 220 shared by all the ports on the PEU controls when the CODEC 216 puts its data onto the local bus. The shift register receives the frame sync signal FS for the particular local bus 96 in question from the FPGA 104. All the ports are mapped sequentially onto timeslots of the local bus, so the shift register just starts shifting a bit through it at each frame syn activation. The output of each cell of the shift register acts as the activation signal for the CODEC and is activated when the bit arrives in the cell corresponding to that port.

Digital data can be sent to the extension phone by the microcontroller if the phone is so equipped, and digital data may also be sent by the phone to the microcontroller via pair. This digital data can include caller ID of the caller and any digital data indicating certain function keys on the phone were pressed.

Referring to FIG. 7 (comprised of FIGS. 7A through 7D), there is shown a flowchart for processing that occurs in the PEU by the DSPs and microcontroller to carry out the commands of PBX call control and driver processes running on host 14. Step 222 happens at initialization and represents the microcontroller loading the program code into RAM in the DSPs from the microcontroller ROM. The microcontroller (hereafter MC) then programs the FPGA 104 by writing programming bits therein. In step 224, the DSP initializes each of its function routines and waits for channel assignment from the MC. The channel assignment is not the channels on the local bus 96 because those are assigned sequentially at startup. The channel assignment is which channels on TDM bus 50 are assigned to carry the data from every port. The PBX process on the host 14 keeps track of which port is assigned to each channel on TDM bus 50. The channel assignment has not yet occurred, but the DSP has started waiting for it.

In step 226, the FPGA listens for the clock on bus 50 and when it detects it, it starts learning its address by watching the match bit of the tokens transmitted to it by the server or other PEUs in the chain.

In step 228, the FPGA interrupts the MC when the FPGA has learned its address. The MC interrupt service routine reads and records the address and sets a flag that the PEU is operational. Each packet traveling in the downstream direction on the packet bus has only a destination address and that destination address can be a broadcast address. The FPGA of the PEU that each packet is addressed to sees an address match and takes the packet off the bus and reassembles it if necessary and writes into a receive FIFO that the MC either constantly polls for new messages or reads in response to an interrupt from the FPGA. The same type process occurs on the switch cards for packets addressed to the host 14.

The MC starts sending a predefined “hello” packet to the PBX process indicating what its PEU address is and how many ports it has in step 230. The PBX process responds in step 232 by assigning a channel on bus 50 to each port of that PEU and sends a packet back to the MC 92 telling it which channels have been assigned to each of its ports. The MC receives the channel assignment packet and parses out the assigned channels among the DSPs on the PEU in step 234. The MC then gives the DSP a start signal to cause it to start all its processing routines.

The processing performed by the DSPs involves routines that: generate and detect DTMF tones and MF DID signaling and call progress tones; do mixing of data from different sources and volume control weighting of the data; play prerecorded messages and recorded voicemail messages out to the ports or CO lines; record of voicemail messages; decoding of caller ID data; echo cancellation and automatic gain control functions. These routines are called either periodically or upon software interrupts in some embodiments. In the preferred embodiment, all processes are performed on a continuous basis as each is called on round robin basis from the main loop during every frame even if there is no data for a particular routine to process. In other words, all the functions that the DSP performs are performed for every channel during every frame. These processes will be discussed sequentially even though they may not occur in the sequence given in the flowchart in some embodiments. In alternative embodiments, each process will be called only when there is data for it to process.

Step 236 represents the beginning of the re-timing and data transfer process and the volume control and mixing process. This is a process which takes data from all 128 timeslots on the bus 126 in FIG. 5 and data from the packet bus directed to various ports and performs a volume control weighting function on the data and mixes it and writes it out onto the appropriate timeslot of the local bus 96. Step 236 represents the process of buffering all 128 timeslots of data from bus 126 in FIG. 5 headed for the ports, and the process of buffering all 32 timeslots of data from local bus 96 headed for the switch cards. The DSP has or is coupled to RAM with a buffer space for each of the 128 inbound timeslots from bus 126 as well as a buffer space for each of the 128 outbound timeslots. Likewise, a buffer space exists in the DSP or in RAM outside the DSP for each of the 32 inbound timeslots and the 32 outbound timeslots of bus 96. The DSP also has a tone buffer for each port to receive data from the MC representing DTMF tones and multifrequency tones for signaling. It also has a buffer for each port it services for voice mail data and outgoing announcement data received from the host via the packet bus.

When data is written into any of the buffers for an outbound timeslot, the circuitry of the DSP automatically writes that data into the corresponding timeslot of the corresponding bus using the frame sync signal for that bus to keep track of the timeslots and get the data into the proper timeslot. The PEUs of the invention are different than the prior art though in that each is coupled to the switch cards by both a packet bus and a TDM bus. The packet bus can carry voicemail message data and prerecorded message data that needs to be played out to a port. In addition, certain call progress tones need to be played out to either the CO line or an extension phone at certain times in a call. For example, when an incoming call comes in on a CO line and the PBX directs it to an extension phone, the extension phone must be caused to ring and “ringback” tone must be sent out the CO port so the caller knows the extension phone is ringing. Likewise, in integrated voice response applications, a prerecorded message giving a menu of options may be played to a user and the user enters a DTMF tone to respond. In such a situation, the user's phone plays the DTMF tones he enters back as a sidetone, but the volume of the outgoing menu message must be kept loud enough so that the sidetone DTMF tones do not drown out the menu message. Likewise, when a user picks up an extension phone and dials out certain DTMF tones, those tones must be played out on the CO line loudly so the CO switch can properly detect what they are.

The PEUs of the invention are unique in that the DSP that services each port can do a volume control function specifically for that port for both the inbound and outbound directions based upon the call situation for that particular port for that particular time. The DSP does this by reading all the timeslot data, announcement or voicemail data and tone data going in each direction, weights each with an appropriate weighting function, sums the resulting data and writes it out the appropriate outbound timeslot's buffer. The process of buffering the outbound and inbound timeslot data for each port is represented by step 236. The process of receiving any tone data from the MC directed to a particular port and buffering it in that port's tone buffer and receiving any incoming tone data from the port and buffering it is represented by step 238. Step 238 also represents the process of receiving the payload sections of data from packets bearing outbound prerecorded message data or voicemail messages and buffering it in an outbound voice buffer and receiving any inbound voice data to be recorded as a voicemail message or to record a conversation in real time is also represented by step 238. Step 240 represents the volume control process of reading the data from each of the sources in each direction, multiplying the data values by a volume control multiplier, and summing the results. Step 242 represents the process of writing the volume control weighted data out to the output buffer that is mapped to the appropriate timeslot on the local bus 96 for traffic out to the port of bus 126 for traffic headed toward the switch card. As soon as the data is written into the output buffer, the hardware of the DSP automatically puts it on the bus in the appropriate timeslot.

Step 244 represents the start of the process of DTMF tone generation wherein the microcontroller receives a packet from the PBX software telling it which DTMF tones to generate for each channel. In step 246, the microcontroller retrieves from its ROM or some other nonvolatile memory the PCM data for the DTMF tones and writes that data into the tone buffer in the DSP corresponding to the channel on which the DTMF tones are to be played. In alternative embodiments, the MC can simply send a message to the DSP telling it which DTMF tones need to be generated on each channel. This is better because the set of possible tones that can be generated is not limited. The DSP can then retrieve the appropriate data that encodes such DTMF tones from its own ROM or other nonvolatile memory and writes into the tone buffer. The DSP then does a volume control weighting function on the tone data with any other data to be played out on the port and writes the resulting sum data into the output buffer for that port which causes the data to be put on the timeslot for that port, as symbolized by step 248.

Step 250 represents the beginning of the play data process that is carried out to play prerecorded announcements, pages, and voicemail messages out to extension phones or to the CO lines. The first thing that happens is a control packet is received by the MC from the PBX software that specifies a particular stream of data is to be played on a particular port, with the particular stream identified by a stream ID.

Step 252 represents the process carried out by the PBX software on the host to broadcast packets on the packet bus 54 to all PEUs every 26 milliseconds. All packets contain PEU destination addresses, but one of the PEU destination addresses is defined as a broadcast address and all PEU FPGAs will pass such a packet on. These packets contain digital data encoding prerecorded voice such as voicemail, outgoing announcements of IVRS systems or live voice pages. The 26 millisecond interval, or some other similarly short interval, is important to prevent noticeable delays in the voice being generated by the CODEC at the port. It is necessary to use a packet size and transmission interval which are short enough such that, considering the latency caused by delays as data propagates through various software layers on the host and through the PEU software and hardware that there be no noticeable delay in the sounds being generated at the port for best performance.

In step 254, the FPGA 104 of every PEU looks at every packet arriving on the packet bus and culls out only those with the PEU addresses which match the address of the PEU or which is a broadcast address. All such packets get copied into the receive FIFO of the MC.

The MC retrieves the packets out of the FIFO in step 256 and looks at the type field. If the type is play data, then the MC looks at the port address. Each packet has a PEU address and a port address. One PEU address is defined for broadcast and one port address is defined for broadcast. If the port address is broadcast, then the MC looks at the stream ID. If the stream ID matches the stream IDs given in any of the previously received control packets, the MC strips out the mu law voice data of the payload section and writes it into an input buffer(s) of the DSP that corresponds to the ports having any matches in the stream ID or IDs given in the previous control packets.

Step 258 is the process that happens if the port address is not a broadcast address. In that case, the MC just strips out the payload data and writes it into the input buffer of the DSP that corresponds to the port identified in the port address of the packet. In alternative embodiments, the voice mail packets can be directly addressed in the port address field to only the ports to which they are being played. In another alternative embodiment, instead of copying the entire payload section into the input buffer of all the ports on which the voice data is to be played, the MC can simply strip out the payload section and write it into shared memory with the DSP and then write a pointer to the data into a pointer store in the DSP for each channel on which the voice data is to be played. Broadcast works best because the play data function is also used to play music on hold or promotional tapes to all CO and extension ports on hold, which in a large system can be a fairly large number of ports.

In step 260, the DSP reads the voice data written into its packet data buffer for each port by the MC and does volume control weighting and mixing with any data off the TDM bus 126 bound for the same port (stored in the DSP TDM buffer for that channel) and any tone data in the tone data buffer, e.g., call waiting tones, to be played on the port and sums the resulting weighted data. The results are then written into the output buffer(s) of all ports on which the voice data is to be played. The particular volume control settings for each direction are set by the PBX software on the basis of requests made by the users by, for example, pushing buttons on her phone.

Step 262 represents the beginning of the recording process. The DSP receives data coming in from the port via the local bus and stores it in a TDM input buffer. TDM data going out to the port is stored in a TDM output buffer. The DSP volume control process reads the TDM data going in each direction in a volume control process for each direction. The volume control process also reads packet data and tone data going in each direction from their respective buffers. Each set of data going in each direction is multiplied by a volume control weighting function and the resulting weighted data is summed. The weighting functions are set by the PBX software on the basis of functions requested by the user.

In step 264, the resulting summed data is written into a record buffer. Once 208 bytes of record data has been accumulated in the record buffer (208 frames), the DSP interrupts the MC and which executes an interrupt service routine. The service routine reads the record buffer data, packetizes it in a packet to be sent to the PBX software and writes it into a transmit FIFO. The FPGA then takes the packet out of the transmit FIFO and transmits it over the packet bus. At the server, the FPGA 46 receives the packet and writes it into a FIFO for the driver software running on the host 14. The driver software takes the packet out of the FIFO and passes it up through the software layers to the process responsible for recording messages or conversations where the payload data is written into a file on the hard disk 16.

The recording and playing of data can be occurring simultaneously.

Step 266 represents the start of the caller ID detection process. The MC is constantly polling the ring detectors 214 in each telephone interface circuit, e.g., 122 in FIG. 1, for every port. In the meantime, the DSP is constantly looking for incoming caller ID data from all ports. Caller ID data is sent as a 1200 baud modem signal from the CO between the first and second ring. When the MC detects the first ring on a CO port, it sets a status in the DSP to start transferring caller ID data to the PBX software.

In step 268, when the DSP has decoded one byte of caller ID data, it interrupts the MC. The MC retrieves the caller ID data and accumulates it by port. When it has received all the CID data for that call, it verifies the checksum. If the caller ID data is correctly received, the MC packetizes it and sends it to the PBX process. The PBX process maintains an object in memory for every call. An object is a small program with an associated data structure that records data defining the attributes of the entity to which the object pertains. The attribute data is updated with the caller ID data, in step 270. The configuration data can be setup to cause the PBX process to automatically route the call to a particular extension based upon the caller ID data without the caller having to dial the extension. Regardless of how the configuration data is setup, the PBX software process always displays the caller ID data by sending it out to the extension phone which the PBX connects to the CO port. This can be done by packetizing the caller ID data and sending it back to the MC with a command in that packet or another packet telling it to send the data to the extension phone via a PCOM digital communication via pair 218 in FIG. 6. In alternative embodiment, the MC may retain the caller ID data in memory after it is sent to the PBX, and the PBX just sends a command packet telling the MC to send that caller ID data to a particular extension phone via the PCOM digital communication interface.

Step 272 represents the process of DTMF tone detection. The DSP is constantly looking for DTMF tones from every port. When DTMF tones are detected, the DSP examines the record buffer and removes these DTMF tones (or mimics thereof) from the record buffer to prevent any DTMF tones appearing in, for example, voicemail recordings or recordings of conversations. This is an optional step, but is preferred since an unintended DTMF tone in a recording can cause unintended results such as deleting the voicemail message at the moment in time when a DTMF tone that means delete in the voicemail context occurs even if the user has not fully listened to the message. This process is done the same way as it was done in the legacy VS1 product and as described in U.S. patent application Ser. No. 09/019,193, filed Feb. 5, 1998, which is hereby incorporated by reference.

Step 274 represents the process of the DSP interrupting the MC when each tone starts and when it stops and decoding each tone including the silence between tones. When a particular tone is starting, the DSP interrupts the MC with an interrupt that says for example, “a 1 is starting.” When the tone stops, another interrupt occurs which says “the tone is stopping and nothing is being detected.” The MC responds to these interrupts by retrieving each decoded digits and the representation of silence and packetizing these digits one by one or as a complete sequence and sending them or it to the PBX process in step 276.

Step 278 represents the start of the MF tone detection process. MF tones are like DTMF tones and are used for various signaling functions. For example, on T1 lines, MF tones are used for Automatic Number Signaling which is caller ID on T1 service and for a service that is like Direct Inward Dialing on POTS lines. The DSP interrupts the MC each time an MF tone starts and when it stops. The MC retrieves each digit, and when a complete sequence has been retrieved, it packetizes the sequence and sends it to the PBX process. In alternative embodiments, the MC can packetize each digit and send it individually to the PBX process.

Steps 280 and 282 represent the process of call progress tone detection. The DSP detects which tones are being generated and builds a status word with one bit per tone set. Any time the status word changes, the DSP interrupts the MC which reads the status word and decodes the cadence of the tones to determine where the system is in the call.

Combinations of tones mean different things. For example, a combination of 440 Hz and 480 Hz is ringback, and 440 and 350 signal dial tone and 620 and 480 is a busy signal. Each has its own cadence. Once the status of the call is determined by the MC, a packet indicating the current state of the call is sent back to the PBX process.

Step 284 represents the optional but preferred step of echo cancellation carried out by the DSPs in each PEU. When recording voicemail messages for example, it is not desired to record echoes of DTMF tones generated by the PBX system for prompts while the user is speaking since such DTMF tones in the recorded message can cause unintended results. Telephone lines are not perfectly terminated always, and an outgoing DTMF prompt can be reflected off an impedance mismatch and come back to the PEU while the caller is recording his message and get into the recorded data. Step 284 prevents this from happening. Removing DTMF tones from recorded data was done in the legacy VS1 system and the details thereof are incorporated by reference. Basically, the DSP uses known algorithms to implement adaptive filters set to filter out DTMF tones.

Step 286 represents the process of performing automatic gain control carried out by the DSP. This is done to normalize variations between line quality and is done using known algorithms in a known way.

Step 288 represents the end of the main loop and return to the top of the loop.

Referring to FIG. 8, there is shown a flowchart of the process the MC carries out to interface with the telephone line interface circuits such as 122 in FIG. 1. Step 287 represents the process of reading configuration data sent to the MC from the PBX at startup time and setting DID switch 202 (FIG. 6) in the telephone interface 122 (FIG. 1) as appropriate. Each port has a switch 206 that is manually manipulated by the installer when the system is set up that tells whether a port is coupled to a CO line or an extension phone. The MC has a process which periodically reads this switch to determine the configuration of each port. This process sends a configuration packet for each port to the DSP which records that data. In alternative embodiments, the DSP need not send back the configuration data packets to the MC because it already knows the configuration of the ports. The DID switch must be in a position to couple the current feed circuit 200 to the transformer 204 when the port is coupled to a CO line to support DID signaling. In certain forms of DID, the PBX signals that it is ready to receive the call by sending loop current back to the CO with the tip voltage polarity reversed from negative to positive 48 volts relative to ring by tip and ring reversal circuit 186. If the port is coupled to an extension phone, switch 202 is set by the MC so that the current feed circuit is not coupled to the transformer.

Step 289 represents the process of reading the configuration data sent by the PBX process indicating whether ground start or loop start signaling protocol is in effect and sending a command to switch 190 in FIG. 6 to send it appropriately. If the ground start signaling protocol is in effect, the MC sends a command to set switch 190 to connect the ground start signaling circuit 188 to connect it to bus 210 so that the ground start protocol signals can be detected. The MC reads the status of these ground start protocol signals via bus 212.

Step 290 represents the process of polling the loop current detector 196 in FIG. 6 every 10 milliseconds. If the same status is found on 2-3 consecutive polls, then the status has been “debounced” and that will be considered to be the new loop current status. In alternative embodiments, interrupts or any other communication mechanism can be used to get the status to the MC.

Step 292 represents the process of polling the ring detector 214. The status is debounced by sampling every 2 msec for multiple 70 msec intervals (70 msec exceeds the slowest ring period by 3 msec). If two consecutive 70 msec periods show a ring signal present (as determined by a count of the number of samples above a certain magnitude greater than a threshold), a ringing status is assumed and reported to the PBX by a packet.

Step 294 represents the PCOMM process of sending digital data such as caller ID data to extension phones capable of receiving and transmitting digital data over bus 218 in FIG. 6. Data to be sent to the phones is collected from the payload sections of packets sent by the PBX process. Data going to the phone is transmitted to the phone over bus 218 during times when the phone is not being polled for data to be sent from the phone. Bus 218 is only half duplex, so reception of data from the phones, such as which function keys were pressed, is done by a polling process. Each phone known to the MC to be operable and PCOM capable (regular analog only phones may also be connected to the ports) is individually addressed (a single character that the phone understands to be a poll) and that poll acts as a permission to send. If a phone has data to send back, it does so in response to the poll. If not, it sends back an “ack, no data” message to indicate it is still up but has no data. In alternative embodiments, the bus 218 may be full duplex and be a TDMA bus or packet bus with the MC buffering data from all the phones sent either during their respective timeslots or via packets addressed to the MC.

Step 296 represents a process of monitoring PCOM status and reporting it to the PBX. If a phone does not respond to its poll for N consecutive polling periods, its PCOM status is marked as down in a table maintained by the MC and reported periodically (or every time the table changes in some embodiments) to the PBX process by one or more packets. Another process in the MC tries to reestablish contact with the downed phones every second to see if they have come back up. If they have, their status is marked as “up” in the table. In the preferred embodiment, the table is reported on a port by port basis.

Step 298 represents a process which executes periodically to read the manual switch 206 on every port to determine if it is coupled to a CO line or to an extension phone. If any change has been made to a port, a packet is put together by the MC and sent to the PBX process in step 300. The status of a port determines what happens when the command on line 194 from the MC is activated in response to a command packet from the PBX process. If the port is coupled to a CO line, when a packet arrives from the PBX saying “go offhook, line 194 is activated and this causes the hookswitch or ring signal circuit 192 to close the loop thereby causing an off hook status. If the port is coupled to an extension phone, when a command packet saying “ring the extension phone” is received from the PBX, line 194 is activated thereby causing circuit 192 to send a ring signal to the extension phone at the cadence specified in the control packet sent from the PBX.

Step 302 represents the process of monitoring all CO ports configured by DID for the existence of an incoming DID call. This is done by the MC by monitoring the loop current detector's status and the ring detector's status. When the CO is ready to send a DID call, it sends loop current but does not send the ring signal. When the MC detects this status, it sends a command to the tip and ring reversal circuit 186 to cause it to reverse the polarity of the tip voltage to +48 volts relative to ring in step 304. This signals the CO that the PBX is ready to receive the digits of the dialed number. The MC then sends a control packet to the PBX indicating there is a DID call inbound.

Step 306 represents the process of the CO sending the dialed digits as DTMF tones. The DSP detects each DTMF tone and interrupts the MC which reads each tone as it starts and packetizes the digit the tone represents and sends it to the PBX process.

Step 308 is the PBX process receiving the dialed digits and mapping them to the extension phone assigned to that dialed number. The PBX process then sends a command packet addressed to the appropriate PEU and the appropriate port saying “ring port x.” In step 310, the MC activates line 194 in FIG. 6 to cause the ring signal circuit 192 to start sending the ring signal to the extension phone named in the control packet at the cadence named in the control packet.

In step 314, the PBX process sends back a control packet to the MC instructing it to answer the call and sends the proper control signals to the proper switch card(s) to set up the switch connection between the CO port and the extension port.

To answer the call, the MC deactivates the Reverse command on line 198 in FIG. 6 to cause the tip and ring reversal circuit 186 to reverse the polarity on the tip and ring lines to signal the CO that the call has been answered by the extension phone, as symbolized by step 316.

In step 318, the CO stop sending ringback tone to the DID caller and connects the analog signals from the caller to the PBX where they are coupled to the extension phone. Steps 320 and 322 represents the process of the analog signals arriving at the port of the PEU coupled to the CO line, digitizing it and putting the data on the timeslot of the TDM bus 50 assigned to that CO port. At the switch card, this data gets transferred to the timeslot assigned to port of the called extension and gets sent back to the PEU having that port where it gets put on the timeslot of the local TDM bus 96 assigned to the port of the called extension. The data is then converted back to an analog signal and output to the extension on pair 180. Incoming analog signals from the called extension make the reverse trip to the CO port in the same way using the timeslots on local bus 96 and bus 50 assigned to the called extension and the CO port.

PEU Processing for an Outgoing Call

Referring to FIG. 9, there is shown a flowchart of the PEU processing for an outgoing phone call placed from an extension phone coupled to the PBX. When an extension phone goes offhook, the regular polling of the loop current detectors of all ports by the MC detects the changed status and reports it to the PBX by a control packet in step 324. In step 326, the PBX process responds by sending a control packet to the MC requesting the MC to play dial tone to the off hook extension. In step 328, the MC writes PCM data for a dial tone into the tone buffer of the DSP assigned to the offhook extension. In step 330, the DSP does the volume control mixing process on the dial tone data and sends it out to the codec for conversion to dial tone analog signals.

In step 332, the DSP detects DTMF digits dialed on the extension phone and interrupts the MC in the manner described above. The MC retrieves the digits and sends them to the PBX process which responds to receipt of the first one by sending a control packet to the MC requesting cessation of the dial tone.

Step 334 represents the process carried out in the PBX process or determining the dialed number and looking up whether it is a local extension or requires a connection to the CO.

If the dialed number is local, the PBX process sends a command packet telling the MC to ring the dialed extension in step 336. The MC responds by activating line 194 in FIG. 6 for the appropriate port to cause the ring signal to be sent. In some embodiments, the cadence is different for inside calls as opposed to calls from the outside. The MC then monitors the status of the loop current detector for that port to determine when and if the extension goes offhook.

In step 338, the MC reports offhook status to the PBX process when loop current is detected. The PBX process responds by setting up the switch connections and data starts transferring in both directions between the extension phones in the appropriate timeslots.

If the dialed number requires a connection to the CO switch, processing vectors out of step 334 to step 340. There, the PBX process optionally does least cost routing and optionally does toll restrictions. Least cost routing is the process of picking a CO connection that will be cheapest for the destination of the call. For example, some companies have leased lines to their subsidiaries in other states. Other companies have different rates for T1 service than POTS service and one or the other may be cheaper for certain destinations. Likewise, different CO lines may have different long distance carrier with different intralata, interlata or out-of-state rates at different times of day so the PBX process takes all factors into consideration and picks the port with the cheapest service at that time for that called destination. Toll restrictions can be applied to certain extensions such as lobby or conference phones to prevent abuse by making unauthorized long distance calls.

The MC responds by activating line 194 to cause the hook switch circuit 192 to take the CO port offhook in step 342. The MC continues to monitor the loop current detector 196 via polling to determine when the CO has gone off hook as indicated by the presence of loop current. When the CO goes offhook, this status is reported to the PBX process.

The CO responds with dial tone, and in step 344, the DSP detects the dial tone and interrupts the MC. The MC reports the dial tone to the PBX via a control packet.

In step 346, the PBX process responds to the dial tone by sending a control packet with a dial string to the MC. The dial string includes the number dialed and may include the area code, a 9 prefix, a pause, prefix numbers, 10-10-xxx numbers, credit card numbers and long distance carrier access numbers. The MC responds in step 348 by writing PCM data for numbers in dial string into appropriate tone buffer of appropriate DSP.

In step 350, the DSP responds by mixing the tone data with any other data to be played out on the CO port and doing volume control adjustments. The resulting data is sent to the codec of the CO port and played to the CO as an analog signal. The MC reports completion of dialing in step 352 to the PBX. The PBX process responds by writing the appropriate switching bits to couple the extension port to the CO port.

In step 354, the CO sends a ringback tone to the PEU. This ringback is detected by the DSP which interrupts the MC. The MC interprets the tone and sends a status packet to the PBX saying ringback detected. The ringback tone and any conversation after the dialed phone is picked up or any voicemail prompts are heard by extension.

Referring to FIG. 10, there is shown a flowchart of the processing of a TEU for incoming T1 calls. A TEU serves the same function as a PEU but for T1 and ISDN digital service port or ports. In the claims, the phrase “port expansion unit” will be used to collectively refer to both PEUs and TEUs. The physical circuitry of a TEU for T1 ports and ISDN ports is the same and the messaging protocol between the TEU and the PBX process is the same for both T1 and ISDN calls. The difference between T1 and ISDN services is in how the TEU detects incoming calls and does signaling with the CO. A T1 line has 24 timeslots per frame with 8000 frames per second with 8 bits per timeslot. Signaling between the CO and the PBX is accomplished by stealing some of the bits in each frame of a channel's timeslots to signal such things as offhook, ringing, etc. In ISDN, there are 24 timeslots per frame and 23 timeslots of every frame are used to send digitized voice data and the 24th timeslot of every frame is used to send the signaling bits pertaining to all the other channels. There are two different types of ISDN, one of which is referred to as PRI. The PBX system disclosed herein supports PRI. The control of the MC, in the preferred embodiment, is done by commercially available software called TS Link 3 and Multi PRI with the Euro ISDN ETSI PRI and Robbed Bit T1 upgrades. This software may be obtained from Telesoft International in Collierville, Tenn. The flowcharts given below for T1 inbound and outbound calls is an alternative embodiment.

Typically, each TEU can service two T1 ports or ISDN ports. The TI/ISDN interface chips 98 strip out the signaling bits from the T1 or ISDN frames coming in from the CO and puts them on signal bit lines which are monitored by the MC 94 in FIG. 1. Step 360 in FIG. 10 represents the beginning of the loop start signaling protocol. Step 360 is the process of the MC constantly polling the interface chips 98 to read the signal bit lines for changes in status. If a ground start protocol is being used, step 380 is performed wherein the MC detects ringing by polling the signaling bits output by the T1/ISDN interface circuit 98. If a wink start protocol is in use, the MC polls the signaling bit outputs of the T1/ISDN interface circuit 98 to detect an offhook condition in step 400.

Step 362 represents the process by the TEU of sending the incoming connection control packet to the PBX in the loop start protocol. This packet tells the PBX that there is an incoming T1 call that is ringing and includes any dialed characters that were detected by the DSP. This packet is accumulated by the MC during a short wait after detection of ringing. T1 loop start protocols do not include DNIS (direct inward dialing) signaling and are simply routed to a predetermined port defined by the configuration data. This predetermined port may be a high speed modem or an auto attendant. If it is an auto attendant, prerecorded announcement data will be played out the T1 port channel with the incoming call asking the person calling to dial the extension of the person they wish to reach. The dialed DTMF tones will be digitized and the DSP in the TEU will detect them and decode them into the characters dialed. The start and end of each DTMF tone will cause the DSP to interrupt the MC, and, in response to the first interrupt, the MC will read the decoded dialed digit from the DSP as described above. The MC will then include the dialed digit in a control packet sent to the PBX process. If there are no dialed digits, the incoming connection request packet generated and sent in step 362 will simply say, “there is an incoming connection request on T1 port x.” If there are no dialed digits, the PBX process then looks up a predetermined port established by configuration data. If there are dialed digits, the PBX process maps the dialed digits to a port having that extension number in step 364. The PBX process then sends a control packet addressed to the correct PEU and port requesting the MC of that PEU to ring the named extension. The MC then activates control line to the ring signal circuit to ring the extension.

The MC then monitors whether the extension goes offhook via the continuous polling of the loop current detection circuit for the port in step 366. When loop current is detected, the MC generates a control packet advising the PBX process that the extension went offhook.

In step 368, the PBX process reacts to the extension going offhook by sending a “connection response” packet back to the MC and then writes bits to the switch card to cause it to set up a connection between the T1 port and the extension. The “connection response” packet causes the MC to write signaling bits to the outbound signal bit lines of the T1/ISDN interface circuit 98 telling the CO that the call is answered and to send any data it has and the call progresses, as symbolized by block 370.

If there were no dialed digits and the predetermined port or extension was an auto attendant, the auto attendant plays a prerecorded announcement out the T1 channel in the same manner as previously described. Basically, the data of the prerecorded announcement is packetized and sent over the packet bus 54 to the TEU where it is received by the MC and written into the voice buffer of the DSP handling the channel on which the connection is occurring. The DSP does volume control mixing and writes the data out to the appropriate output buffer mapped to the timeslot on the local bus 134 mapped to the T1 channel in use. The T1/ISDN interface circuit then puts it out on the outbound timeslots of the channel in use. The outbound message will prompt the user to dial the extension of the called party or some other extension. These DTMF tones will be decoded by the DSP 144 for the channel and passed to the MC which passes them to the PBX via a control packet. The PBX then maps them to the extension and sets up the call in the manner previously described (ring phone, monitor offhook, advise CO of connection made status and set up switch connection).

Returning to the consideration of the ground start protocol, step 380 detects ringing in the known signaling manner of the ground start protocol. After ringing is detected, the processing is the same as steps 362, 364, 366, 368 and 370 for the loop start protocol.

The wink start protocol for incoming T1 digital service starts with step 400 where the MC detects offhook status in the signaling bits. Wink start protocol supports DNIS (the T1/ISDN equivalent of DID) and ANI (the T1/ISDN equivalent of caller ID). Any DNIS data and ANI data will be sent as DTMF digits grouped in two fields of a string that starts and stops with an asterisk and has the two fields separated by an asterisk. ANI data will be in the first field and DNIS data will be in the second. If either field is blank, there will simply be two asterisks next to each other, but there are always three asterisks. Wink start also supports dialing manually.

The MC responds to the detection of offhook status by waiting 120 milliseconds to effectively debounce the offhook status and see if the CO is still off hook, and then sending back a “wink,” as symbolized by step 402. A wink is defined as transmission back to the CO for a short interval of control bits that indicate offhook status. The duration of the wink is set by the configuration data.

In step 404, the MC waits for the end of the wink and then starts looking for ANI and DNIS digits. As digits arrive, they are accumulated until one second passes without receiving any further digits. In step 406, the MC collects all the DNIS and ANI digits collected during the previous dialing timer intervals and packetizes them and transmits the packet to the PBX process.—From there, the process is the same as described in steps 364, 366, 368 and 370.

Referring to FIG. 11, there is shown a flowchart of the processing of an alternative embodiment for placement of an outbound call on a T1 port. Step 420 represents the process of the extension phone going offhook and this fact being detected by the MC in its regular polling of the loop current detector. The MC reports this offhook status to the PBX process.

In step 422, the DSP detects any DTMF tones of dialed digits and decodes them. The DSP interrupts the MC which retrieves each dialed digit and packetizes it (or collects them all and packetizes the group) and sends them to the PBX process.

The PBX maps the dialed digits either to a local extension or to an outside line and, in some embodiments, does least cost routing. For purposes of this discussion, assume that the least cost routing process indicates that the T1 service is the cheapest. In that case, the PBX process sends a command packet of the TEU coupled to the Ti port to be used (the PBX picks an unused channel of a T1 port that does not have all channels in use) and addressed to a particular channel of a particular T1 port requesting the MC to make a connection request. This packet also includes a dial string.

What happens next depends upon the protocol in use. In the case of ground start, the MC sends ground ring signaling bits to the CO on the outgoing robbed bits in step 428. In the case of loop start protocol, the TEU MC sends “loop closed” signal bits in the outgoing robbed bits in step 450. In step 452, the MC either waits a predetermined time for dial tone from the CO and assumes it is there at the end of that time or the MC actually detects dial tone in the incoming robbed bits. In some embodiments, the MC also sends a control packet indicating the presence of dial tone to the PBX process.

In step 454, the MC copies PCM data for the dial string into the appropriate tone buffer of the DSP that is handling the outbound channel of the particular T1 port upon which the call is to be made. In step 456, the DSP does volume control mixing of the tone data with any other data to be played out on the channel, and the resulting volume adjusted data is written into the output buffer corresponding to the channel to be used for the phone call. When the data is written into the output buffer, the DSP hardware automatically puts it onto the correct timeslot mapped to the channel and to that output buffer.

In step 458, the MC sends back a “dial complete” control packet, and the PBX process responds by setting up the switch connection between the extension port and the appropriate channel of the appropriate T1 port.

Returning to the consideration of the ground start protocol, step 430 represents the process of the MC waiting for the “current feed” signal bits to arrive from the CO. The MC responds to the “current feed” signal bits in step 430 by sending “loop closed” signal bits to the CO indicating to the CO that the MC is ready to send the dial string. The rest of the call is then completed as described above in steps 454, 456, 458, 460 and 462.

Turning to consideration of the wink start protocol, step 470 represents the process of sending the offhook signal bits to the CO to initiate the connection process. The CO responds by sending a “wink”, i.e., a momentary transmission of off hook signal bits to the PBX. The MC detects the wink signal bits in step 474 and waits a short interval and then starts dialing the dial string by the process of step 454. The rest of the call is then completed as detailed in steps 456, 458, 460 and 462, as symbolized by step 476.

The process carried out by the system in handling incoming and outgoing PRI digital service calls is as embodied in the software previously referred to from Telesoft.

Referring to FIG. 12, there is shown a flowchart of the processing of the DSPs 42 on the switch cards 28. These DSPs do conferencing. Conference calling was done on the prior art VS1 system available from the assignee, but it was done on the switch chips in those systems. The DSPs 42 do conference calling along with volume control and automatic gain control functions and they are the first DSPs so programmed in a distributed DSP personal computer based PBX system such as described herein.

There are 128 channels on the TDMA bus 50. The last 32 of these are used for conferencing by the DSPs 42 and the remaining 96 channels are passed through unmodified in each direction. The DSPs such as 42 perform echo canceling on the conference channels, perform automatic gain control and volume control in mixing the data from the different channels together to generate the combined sound that all conference participants hear. DSP 42 is actually three DSPs which are daisy chained in the preferred embodiment since one DSP does not have enough processing power to do echo canceling on all 32 channels. Thus, the echo canceling function is split among the 3 DSPs with the first one doing any known echo canceling routine on the outputs from the switch 36 for the first 13 channels and passes the other channels on to the second DSP untouched, as symbolized by step 472. The second DSP in the chain then does a known echo canceling routine on the next 13 of the 32 conferencing channels and passes the first 13 and the last 6 on to the third DSP, as symbolized by step 474. The third DSP then does echo canceling on the last 6 conferencing channels and passes all the channel data on to the FPGA or the switch chips 36, for transmission on the TDMA bus 50 or switching to another timeslot depending upon direction of travel of the data. Each of the three DSPs has a separate input buffer and output buffer for each of the 128 channels going in each direction. The echo canceling process goes on for the last 32 channels in each direction.

The third DSP in the chain in each direction does automatic gain control on all 32 of the conference channels after it has done echo cancellation on its 6 channels and has received the echo cancelled data from the other 26 conference channels from the first two DSPS, as symbolized by step 476. The automatic gain control function is performed such that each conference participant hears the other participants at approximately the same volume despite the differences in volume of the raw data arriving from the different CO POTS or T1 lines.

In step 478, the third DSP uses conference setup data written into its memory by the PBX process to determine which channels belong to which conferences.

In step 480, the volume control process is started. This is done for each conference by calculating the sum of all data from the timeslots assigned to the conference plus any beep tone data stored in the tone data buffer for the conference. The beep tone is used to advise all conferees that a new conferee has joined the conference.

In step 482, the process of calculating the input data that will be put into each channel's timeslot, i.e., the data that will be converted into the sound that the conferee assigned to that channel will hear, is started. This is a two step process. The general idea is to allow each conferee to hear all the other participants but not his or her own voice, and then to adjust the resulting data by the overall conference volume. The first step is symbolized by step 482 where the conferees own voice data, i.e., the “channel output,” is subtracted from the conference sum. The second step is to take the resulting data for each channel and to multiply it by the conference volume, as symbolized by step 484. The conference volume is specified in the DSP memory by the PBX process.

In step 486, the third DSP writes the calculated conference input data for each channel and writes it into its output buffer assigned to that channel so that it will get passed to DSP 2 and thence to DSP1. Data from all other channels is passed through unchanged. DSP 2 and DSP 1 pass all data from all channels including conference channels through unchanged.

Although the invention has been described with reference to the preferred and alternative embodiments disclosed herein, those skilled in the art will appreciate various modifications and enhancements that do not depart from the spirit and scope of the teachings of the invention. For example, in some alternative embodiments, the status signals sent from the microcontrollers to the PBX process or host computer can be sent over one or more channels on the TDMA bus 50 which are dedicated to control and management information or they may be sent by a “robbed bit” method similar to T1 service signaling via robbed bits. Likewise, commands sent back from the PBX process to the microcontrollers in the PEUs and TEUs can be sent via one or more dedicated control channels on the TDMA bus 50 or via robbed bits from channels carrying voice data. All such modifications and enhancements are intended to be included within the scope of the claims appended hereto. 

1. An apparatus, comprising: a personal computer having a plurality of expansion slots; one or more switch card circuits coupled to said expansion slots, each including a switching integrated circuit capable of coupling data in at least one timeslot to at least one other timeslot on a time division multiplexed bus and further comprising an interface circuit for a time division multiplexed bus and a packet switched bus; and one or more port expansion units coupled together in a daisy chain by a time division multiplexed bus and a packet switched bus, each said daisy chain of port expansion units having a port expansion unit in the chain coupled to a switch card via a time division multiplexed bus and a packet switched bus which are extensions of the packet switched and time division multiplexed buses which couple each daisy chain together and also having a bus interface circuit for said buses, each said port expansion unit including one or more port interface circuits for interfacing the port expansion unit to a telecommunication line, one or more digital signal processors coupled between said bus interface circuit and said port interface circuits, and one or more microcontrollers coupled between said bus interface circuit and said one or more digital signal processors, wherein said personal computer includes a private branch exchange (PBX) program, which when executed causes said personal computer to implement a PBX process to communicate with said one or more microcontrollers via port interface circuits on each said port expansion unit to determine call progress and port status, to send data to said microcontrollers to control signaling functions implemented by said port interface circuits, and to control switching of said switch cards to implement at least some conventional PBX functionality.
 2. The apparatus of claim 1, wherein said microcontrollers in each said port expansion unit are programmed to read status signals from said port interface circuits, to receive commands from said PBX process, to send commands to said port interface circuits to implement signaling functions, and to send write tone data to a digital signal processor in said port expansion unit to cause tone generation on channels designated in data received from said PBX program for signaling purposes.
 3. The apparatus of claim 2, wherein said one or more digital signal processors in said port expansion unit are programmed to receive data from said time division multiplexed bus for channels corresponding to PBX connections and said tone data, and to implement a volume control weighting function to mix said channel data and said tone data to form resulting data, and to send the resulting data to the appropriate port interface circuit for transmission to a central office.
 4. The apparatus of claim 2, where said one or more digital signal processors are programmed to receive data from port interface circuits, to place said data onto predetermined channels of said time division multiplexed bus, to receive data from said port interface circuits representing each incoming signaling tone, and to decode said data to determine which tone has been received to interrupt said microcontroller for each said tone received.
 5. The apparatus of claim 4, said one or more microcontrollers being programmed to respond to each interrupt from said one or more digital signal processors, to determine the tone which has been received, and to send a message to said PBX process identifying the tone which has been received.
 6. The apparatus of claim 1, said personal computer including programming, which when executed as part of a telephony process including voice mail, to send data representing voice mail messages to the appropriate one or more ports via packets on said packet switched bus, wherein the microcontrollers that receive said packets are programmed to strip out the payload data of each said packet and write a resulting packet into a packet data buffer accessible by said one or more digital signal processors.
 7. The apparatus of claim 6, wherein said one or more digital signal processors are programmed to mix said packet data in with tone data and said channel data in a volume control weighting process to generate combined data and to send the combined data out to the appropriate one or more said ports for transmission to a central office.
 8. The apparatus of claim 1, wherein said one or more digital signal processors are programmed to receive data to be recorded as a voicemail message, to save said data into a record buffer, to interrupt at least one of said one or more microcontrollers, the one or more microcontrollers in said port expansion unit to receiving said interrupt, to retrieve said voicemail data from said record buffer, to packetize said voicemail data, and to transmit, via said packet switched bus, said packetized voicemail data to a telephony process on said personal computer.
 9. The apparatus of claim 8, said personal computer including programming, which when executed as part of a telephony process including voice mail, causes the personal computer to receive said packetized voicemail data including voicemail messages to be recorded from one or more of the port expansion interface units.
 10. A method of operating a PBX system on a personal computer, comprising: loading code for one or more digital signal processors (DSPs), and programming a field programmable gate array (FPGA) by a microcontroller (MC); initializing, by said one or more DSPs, each function and waiting for channel assignments by said MC; learning an address of the FPGA by watching a match bit of a token; interrupting, by the FPGA, the MC after the address has been learned, and setting an operational flag, transmitting a packet to a PBX program on a host computer to indicate a port expansion unit (PEU) including the MC, the one or more DSPs, and the FPGA is operational; transmitting a predefined packet to said PBX program indicating how many ports the PEU includes; receiving a response back from said PBX program with an assignment of a channel on a bus for each port on the PEU; distributing, by the microcontroller, of the assigned channels among the one or more DSPs; and transmitting to the one or more DSPs a start processing signal.
 11. The method of claim 10, further including the one or more DSPs buffering first timeslot data for a plurality of timeslots from a bus that is to be transmitted to said ports of the PEU and buffering second timeslot data for a plurality of timeslots of incoming data from said ports of the PEU.
 12. The method of claim 11, further including the one or more DSPs receiving and buffering in one or more tone buffers tone data received from the MC.
 13. The method of claim 12, further including the one or more DSPs receiving payloads of voice data from prerecorded messages and voicemail packets which were transmitted over a packet bus.
 14. The method of claim 13, further including the one or more DSPs reading the first timeslot data, the second timeslot data, the tone data, and the voice data and weighing data according to a volume control weighting function and summing the results to created resulting volume control mixed data.
 15. The method of claim 14, further including the one or more DSPs writing the resulting volume control mixed data to an appropriate buffer.
 16. The method of claim 11, wherein the microcontroller receives a host packet from the PBX program identifying dial tone multi-frequency (DTMF) tones to generate for each channel.
 17. The method of claim 16, said microcontroller retrieving pulse-code modulated (PCM) data for the identified DTMF tones and writing the PCM data into the one or more tone buffers corresponding to each channel.
 18. The method of claim 17, said one or more DSPs performing volume control weighting on the PCM data in the one or more tone buffers, summing the volume control weighted data, and writing the summed volume control weighting data into an output buffer corresponding to each said channel. 